1a018dcda3
At the top level, setting a track on an RtpSender is equivalent to setting a source (previously called a renderer) on a voice send stream. An RtpSender without a track is not supposed to send data (not even muted data), so a send stream without a source shouldn't send data. Also replacing SendFlags with a boolean and implementing "Start" and "Stop" methods on AudioSendStream, which was planned anyway and simplifies this CL. R=pthatcher@webrtc.org, solenberg@webrtc.org Review URL: https://codereview.webrtc.org/1741933002 . Cr-Commit-Position: refs/heads/master@{#11918}
WebRTC is a free, open software project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose.
Our mission: To enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.
The WebRTC initiative is a project supported by Google, Mozilla and Opera, amongst others. This page is maintained by the Google Chrome team.
Development
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.
More info
- Official web site: http://www.webrtc.org
- Master source code repo: https://chromium.googlesource.com/external/webrtc
- Samples and reference apps: https://github.com/webrtc
- Mailing list: http://groups.google.com/group/discuss-webrtc
- Continuous build: http://build.chromium.org/p/client.webrtc